[Asterisk-code-review] chan rtp.c: Simplify options to UnicastRTP channel creation. (asterisk[master])
Joshua Colp
asteriskteam at digium.com
Wed Jun 8 05:13:59 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: chan_rtp.c: Simplify options to UnicastRTP channel creation.
......................................................................
chan_rtp.c: Simplify options to UnicastRTP channel creation.
Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])
Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.
Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
---
M CHANGES
M channels/chan_rtp.c
2 files changed, 69 insertions(+), 10 deletions(-)
Approvals:
Joshua Colp: Looks good to me, approved; Verified
Matthew Fredrickson: Looks good to me, but someone else must approve
diff --git a/CHANGES b/CHANGES
index 608a4a4..e799f71 100644
--- a/CHANGES
+++ b/CHANGES
@@ -135,6 +135,32 @@
seconds. Setting this to a higher value may help in lagged networks or those
experiencing high packet loss.
+chan_rtp (was chan_multicast_rtp)
+------------------
+ * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.
+
+ * The format for dialing a unicast RTP channel is:
+ UnicastRTP/<destination-addr>[/[<options>]]
+ Where <destination-addr> is something like '127.0.0.1:5060'.
+ Where <options> are in standard Asterisk flag options format:
+ c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+ e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
+
+ * New options were added for a multicast RTP channel. The format for
+ dialing a multicast RTP channel is:
+ MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
+ Where <type> can be either 'basic' or 'linksys'.
+ Where <destination-addr> is something like '224.0.0.3:5060'.
+ Where <control-addr> is something like '127.0.0.1:5060'.
+ Where <options> are in standard Asterisk flag options format:
+ c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+ i(<address>) - Specify the interface address from which multicast RTP
+ is sent.
+ l(<enable>) - Set whether packets are looped back to the sender. The
+ enable value can be 0 to set looping to off and non-zero to set
+ looping on.
+ t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
+
chan_sip
------------------
* New 'rtpbindaddr' global setting. This allows a user to define which
diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
index 0936028..0fe66bd 100644
--- a/channels/chan_rtp.c
+++ b/channels/chan_rtp.c
@@ -176,7 +176,7 @@
fmt = ast_format_cap_get_format(cap, 0);
}
if (!fmt) {
- ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
+ ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
args.destination);
goto failure;
}
@@ -230,6 +230,25 @@
return NULL;
}
+enum {
+ OPT_RTP_CODEC = (1 << 0),
+ OPT_RTP_ENGINE = (1 << 1),
+};
+
+enum {
+ OPT_ARG_RTP_CODEC,
+ OPT_ARG_RTP_ENGINE,
+ /* note: this entry _MUST_ be the last one in the enum */
+ OPT_ARG_ARRAY_SIZE
+};
+
+AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
+ /*! Set the codec to be used for unicast RTP */
+ AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
+ /*! Set the RTP engine to use for unicast RTP */
+ AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
+END_OPTIONS );
+
/*! \brief Function called when we should prepare to call the unicast destination */
static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
@@ -240,11 +259,13 @@
struct ast_channel *chan;
struct ast_format_cap *caps = NULL;
struct ast_format *fmt = NULL;
+ const char *engine_name;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(destination);
- AST_APP_ARG(engine);
- AST_APP_ARG(format);
+ AST_APP_ARG(options);
);
+ struct ast_flags opts = { 0, };
+ char *opt_args[OPT_ARG_ARRAY_SIZE];
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
@@ -262,17 +283,26 @@
goto failure;
}
- if (!ast_strlen_zero(args.format)) {
- fmt = ast_format_cache_get(args.format);
+ if (!ast_strlen_zero(args.options)
+ && ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
+ ast_strdupa(args.options))) {
+ ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
+ args.options);
+ goto failure;
+ }
+
+ if (ast_test_flag(&opts, OPT_RTP_CODEC)
+ && !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
+ fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
if (!fmt) {
- ast_log(LOG_ERROR, "Format '%s' not found for sending RTP to '%s'\n",
- args.format, args.destination);
+ ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
+ opt_args[OPT_ARG_RTP_CODEC], args.destination);
goto failure;
}
} else {
fmt = ast_format_cap_get_format(cap, 0);
if (!fmt) {
- ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
+ ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
args.destination);
goto failure;
}
@@ -283,12 +313,15 @@
goto failure;
}
+ engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
+ opt_args[OPT_ARG_RTP_ENGINE], NULL);
+
ast_ouraddrfor(&address, &local_address);
- instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL);
+ instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
if (!instance) {
ast_log(LOG_ERROR,
"Could not create %s RTP instance for sending media to '%s'\n",
- S_OR(args.engine, "default"), args.destination);
+ S_OR(engine_name, "default"), args.destination);
goto failure;
}
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
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